Method and apparatus for level control in blending an audio signal in an in-band on-channel radio system

ABSTRACT

A method for processing a digital audio broadcast signal includes: separating an analog audio portion and a digital audio portion of the digital audio broadcast signal; determining the loudness of the analog audio portion and the digital audio portion over a first short time interval; using the loudness of the analog and digital audio portions to calculate a short term average gain; determining a long term average gain; converting one of the long term average gain or the short term average gain to dB; if an output has been blended to digital, adjusting a digital gain parameter by a preselected increment to produce a digital gain parameter; if an output has not been blended to digital, setting the digital gain parameter to the short term average gain; providing the digital gain parameter to an audio processor; and repeating the above steps using a second short time interval.

FIELD OF THE INVENTION

The described methods and apparatus relate to digital radio broadcastreceivers and, in particular, to methods and apparatus for levelalignment of analog and digital pathways in digital radio receivers.

BACKGROUND OF THE INVENTION

Digital radio broadcasting technology delivers digital audio and dataservices to mobile, portable, and fixed receivers. One type of digitalradio broadcasting, referred to as in-band on-channel (IBOC) digitalaudio broadcasting (DAB), uses terrestrial transmitters in the existingMedium Frequency (MF) and Very High Frequency (VHF) radio bands. HDRadio™ technology, developed by iBiquity Digital Corporation, is oneexample of an IBOC implementation for digital radio broadcasting andreception.

IBOC technology can provide digital quality audio, superior to existinganalog broadcasting formats. Because each IBOC signal is transmittedwithin the spectral mask of an existing AM or FM channel allocation, itrequires no new spectral allocations. IBOC promotes economy of spectrumwhile enabling broadcasters to supply digital quality audio to thepresent base of listeners.

The National Radio Systems Committee, a standard-setting organizationsponsored by the National Association of Broadcasters and the ConsumerElectronics Association, adopted an IBOC standard, designated NRSC-5, inSeptember 2005. NRSC-5, the disclosure of which is incorporated hereinby reference, sets forth the requirements for broadcasting digital audioand ancillary data over AM and FM broadcast channels. The standard andits reference documents contain detailed explanations of theRF/transmission subsystem and the transport and service multiplexsubsystems. Copies of the standard can be obtained from the NRSC athttp://www.nrscstandards.org/standards.asp. iBiquity's HD Radiotechnology is an implementation of the NRSC-5 IBOC standard. Furtherinformation regarding HD Radio technology can be found atwww.hdradio.com and www.ibiquity.com.

IBOC signals can be transmitted in a hybrid format including an analogmodulated carrier in combination with a plurality of digitally modulatedcarriers or in an all-digital format wherein the analog modulatedcarrier is not used. Using the hybrid mode, broadcasters may continue totransmit analog AM and FM simultaneously with higher-quality and morerobust digital signals, allowing themselves and their listeners toconvert from analog-to-digital radio while maintaining their currentfrequency allocations.

Both AM and FM In-Band On-Channel (IBOC) hybrid broadcasting systemsutilize a composite signal including an analog modulated carrier and aplurality of digitally modulated subcarriers. Program content (e.g.,audio) can be redundantly transmitted on the analog modulated carrierand the digitally modulated subcarriers. The analog audio is delayed atthe transmitter by a diversity delay.

In the absence of the digital audio signal (for example, when thechannel is initially tuned) the analog AM or FM backup audio signal isfed to the audio output. When the digital audio signal becomesavailable, a blend function smoothly attenuates and eventually replacesthe analog backup signal with the digital audio signal while blending inthe digital audio signal such that the transition preserves somecontinuity of the audio program. Similar blending occurs during channeloutages which corrupt the digital signal. In this case the analog signalis gradually blended into the output audio signal by attenuating thedigital signal such that the audio is fully blended to analog when thedigital corruption appears at the audio output. Corruption of thedigital audio signal can be detected during the diversity delay timethrough cyclic redundancy check (CRC) error detection means, or otherdigital detection means in the audio decoder or receiver.

The concept of blending between the digital audio signal of an IBOCsystem and the analog audio signal has been previously described in, forexample, U.S. Pat. Nos. 7,546,088; 6,178,317; 6,590,944; 6,735,257;6,901,242; and 8,180,470, the disclosures of which are herebyincorporated by reference. The diversity delay and blend allow thereceiver to fill in the digital audio gaps with analog audio whendigital outages occur. The diversity delay ensures that the audio outputhas a reasonable quality when brief outages occur in a mobileenvironment (for example, when a mobile receiver passes under a bridge).This is because the time diversity causes the outages to affectdifferent segments of the audio program for the digital and analogsignals.

In the receiver, the analog and digital pathways may be separately, andthus asynchronously, processed. In a software implementation, forexample, analog and digital demodulation processes may be treated asseparate tasks using different software threads. Subsequent blending ofthe analog and digital signals requires that the signals be aligned intime before they are blended.

Both FM and AM Hybrid In-Band On-Channel (IBOC) HD Radio™ receiversrequire an audio blend function for the purposes of blending to the FMor AM analog backup signal when the digital signal is unavailable. Themaximum blend transition time is limited by the diversity delay andreceiver decoding times, and is typically less than one second. Frequentblends can sometimes degrade the listening experience when the audiodifferences between the digital and analog are significant.

Blending will typically occur at the edge of digital coverage and atother locations within the coverage contour where the digital waveformis corrupted. When a short outage does occur, such as traveling under abridge, the loss of digital audio is replaced by an analog signal. Whenblending occurs, it is important that the content on the analog audioand digital audio channels are aligned in both time and level (i.e.,loudness) to ensure that the transition is barely noticed by thelistener. Optimally, the listener will notice little other than possibleinherent quality differences in analog and digital audio at these blendpoints. However, if the broadcast station does not have the analog anddigital audio signals aligned, then the result could be a harsh soundingtransition between digital and analog audio. The misalignment may occurbecause of audio processing differences between the analog audio anddigital audio paths at the broadcast facility. Furthermore the analogand digital signals are typically generated with two separate signalgeneration paths before combining for output. The use of differentanalog processing techniques and different signal generation methodsmakes the alignment of these two signals nontrivial. The blending shouldbe smooth and continuous, which can happen only if the analog anddigital audio are aligned in both time and level.

It would be desirable to process a digital radio signal in a manner thatallows blending of the digital and analog components without an abruptchange in loudness of the audio output.

SUMMARY

In one embodiment, a method for processing a digital audio broadcastsignal includes: (a) separating an analog audio portion of the digitalaudio broadcast signal from a digital audio portion of the digital audiobroadcast signal; (b) determining a loudness of the analog audio portionover a first short time interval; (c) determining a loudness of thedigital audio portion over the first short time interval; (d) using theloudness of the analog audio portion and the loudness of the digitalaudio portion to calculate a short term average gain; (e) determining along term average gain; (f) converting one of the long term average gainor the short term average gain to dB; (g) if an output has been blendedto digital, adjusting a digital gain parameter by a preselectedincrement to produce an adjusted digital gain parameter; (h) if anoutput has not been blended to digital, setting the digital gainparameter to the short term average gain; (i) providing the digital gainparameter to an audio processor; and (j) repeating steps (a) through (i)using a second short time interval.

In another embodiment, a radio receiver includes: processing circuitryconfigured to: (a) separate an analog audio portion of the digital audiobroadcast signal from a digital audio portion of the digital audiobroadcast signal; (b) determine a loudness of the analog audio portionover a first short time interval; (c) determine a loudness of thedigital audio portion over the first short time interval; (d) use theloudness of the analog audio portion and the loudness of the digitalaudio portion to calculate a short term average gain; (e) determine along term average gain; (f) convert one of the long term average gain orthe short term average gain to dB; (g) if an output has been blended todigital, adjust a digital gain parameter by a preselected increment toproduce an adjusted digital gain parameter; (h) if an output has notbeen blended to digital, set the digital gain parameter to the shortterm average gain; (i) provide the digital gain parameter to an audioprocessor; and (j) repeat steps (a) through (i) using a second shorttime interval.

BRIEF DESCRIPTION OF THE DRAWINGS

FIG. 1 is a functional block diagram of an exemplary digital radiobroadcast transmitter.

FIG. 2 is a functional block diagram of an exemplary digital radiobroadcast receiver.

FIG. 3 is a functional block diagram that shows separate digital andanalog signal paths in a receiver.

FIG. 4 is a functional block diagram that shows elements of a timealignment module.

FIG. 5 is a flow block diagram of a method for level alignment inaccordance with an embodiment of the invention.

DETAILED DESCRIPTION OF THE INVENTION

Embodiments described herein relate to the processing of the digital andanalog components of a digital radio broadcast signal. While aspects ofthe disclosure are presented in the context of an exemplary IBOC system,it should he understood that the present disclosure is not limited toIBOC systems and that the teachings herein are applicable to other formsof digital radio broadcasting as well.

Referring to the drawings, FIG. 1 is a block diagram of an exemplarydigital radio broadcast transmitter 10 that broadcasts digital audiobroadcasting signals. The exemplary digital radio broadcast transmittermay be a DAB transmitter such as an AM or FM IBOC transmitter, forexample. An input signal source 12 provides the signal to betransmitted. The source signal may take many forms, for example, ananalog program signal that may represent voice or music and/or a digitalinformation signal that may represent message data such as trafficinformation. A baseband processor 14 processes the source signal inaccordance with various known signal processing techniques, such assource coding, interleaving and forward error correction, to producein-phase and quadrature components of a complex baseband signal on lines16 and 18, and to produce a transmitter baseband sampling clock signal20. Digital-to-analog converter (DAC) 22 converts the baseband signalsto an analog signal using the transmitter baseband sampling clock 20,and outputs the analog signal on line 24. The analog signal is shiftedup in frequency and filtered by the up-converter block 26. This producesan analog signal at an intermediate frequency f_(if) on line 28. Anintermediate frequency filter 30 rejects alias frequencies to producethe intermediate frequency signal f_(if) on line 32. A local oscillator34 produces a signal f_(lo) on line 36, which is mixed with theintermediate frequency signal on line 32 by mixer 38 to produce sum anddifference signals on line 40. The unwanted intermodulation componentsand noise are rejected by image reject filter 42 to produce themodulated carrier signal f_(c) on line 44. A high power amplifier (HPA)46 then sends this signal to an antenna 48.

In one example, a basic unit of transmission of the DAB signal is themodem frame, which is typically on the order of a second in duration.Exemplary AM and FM IBOC DAB transmission systems arrange the digitalaudio and data in units of modem frames. Some transmission systems areboth simplified and enhanced by assigning a fixed number of audio framesto each modem frame. The audio frame period is the length of timerequired to render, e.g., play back audio for a user, the samples in anaudio frame. For example, if an audio frame contains 1024 samples, andthe sampling period is 22.67 μsec, then the audio frame period would beapproximately 23.2 milliseconds. A scheduler determines the total numberof bits allocated to the audio frames within each modem frame. The modemframe duration is advantageous because it may enable sufficiently longinterleaving times to mitigate the effects of fading and short outagesor noise bursts such as may be expected in a digital audio broadcastingsystem. Therefore the main digital audio signal can be processed inunits of modem frames, and audio processing, error mitigation, andencoding strategies may be able to exploit this relatively large modemframe time without additional penalty.

In typical implementations, an audio encoder may be used to compress theaudio samples into audio frames in a manner that is more efficient androbust for transmission and reception of the IBOC signal over the radiochannel. The audio encoder encodes the audio frames using the bitallocation for each modem frame. The remaining bits in the modem frameare typically consumed by the multiplexed data and overhead. Anysuitable audio encoder can initially produce the compressed audio framessuch as an HDC encoder as developed by Coding Technologies of DolbyLaboratories, Inc.; an Advanced Audio Coding (AAC) encoder; an MPEG-1Audio Layer 3 (MP3) encoder; or a Windows Media Audio (WMA) encoder.Typical lossy audio encoding schemes, such as AAC, MP3, and WMA, utilizethe modified discrete cosine transform (MDCT) for compressing audiodata. MDCT based schemes typically compress audio samples in blocks of afixed size. For example, in AAC encoding, the encoder may use a singleMDCT block of length 1024 samples or 8 blocks of 128 samples.Accordingly, in implementations using an AAC coder, for example, eachaudio frame could be comprised of a single block of 1024 audio samples,and each modem frame could include 64 audio frames. In other typicalimplementations, each audio frame could be comprised of a single blockof 2048 audio samples, and each modem frame could include 32 audioframes. Any other suitable combination of sample block sizes and audioframes per modem frame could be utilized.

In an exemplary IBOC DAB system, the broadcast signal includes mainprogram service (MPS) audio, MPS data (MPSD), supplemental programservice (SPS) audio, and SPS data (SPSD). MPS audio serves as the mainaudio programming source. In hybrid modes, it preserves the existinganalog radio programming formats in both the analog and digitaltransmissions. MPSD, also known as program service data (PSD), includesinformation such as music title, artist, album name, etc. Supplementalprogram service can include supplementary audio content as well as PSD.Station Information Service (SIS) is also provided, which comprisesstation information such as call sign, absolute time, positioncorrelated to GPS, data describing the services available on thestation. In certain embodiments, Advanced Applications Services (AAS)may be provided that include the ability to deliver many data servicesor streams and application specific content over one channel in the AMor FM spectrum, and enable stations to broadcast multiple streams onsupplemental or sub-channels of the main frequency.

A digital radio broadcast receiver performs the inverse of some of thefunctions described for the transmitter. FIG. 2 is a block diagram of anexemplary digital radio broadcast receiver 50. The exemplary digitalradio broadcast receiver 50 may be a DAB receiver such as an AM or FMIBOC receiver, for example. The DAB signal is received on antenna 52. Abandpass preselect filter 54 passes the frequency band of interest,including the desired signal at frequency f_(c), but rejects the imagesignal at f_(c)-2f_(if) (for a low side lobe injection localoscillator). Low noise amplifier (LNA) 56 amplifies the signal. Theamplified signal is mixed in mixer 58 with a local oscillator signalf_(lo) supplied on line 60 by a tunable local oscillator 62. Thiscreates sum (f_(c)+f_(lo)) and difference (f_(c)−f,_(lo)) signals online 64. Intermediate frequency filter 66 passes the intermediatefrequency signal f_(if) and attenuates frequencies outside of thebandwidth of the modulated signal of interest. An analog-to-digitalconverter (ADC) 68 operates using the front-end clock 70 to producedigital samples on line 72. Digital down converter 74 frequency shifts,filters and decimates the signal to produce lower sample rate in-phaseand quadrature signals on lines 76 and 78. The digital down converter 74also outputs a receiver baseband sampling clock signal 80. A basebandprocessor 82, operating using the master clock 84 that may or may not begenerated from the same oscillator as the front-end clock 70, thenprovides additional signal processing. The baseband processor 82produces output audio samples on line 86 for output to audio sink 88.The output audio sink may be any suitable device for rendering audiosuch as an audio-video receiver or car stereo system.

FIG. 3 is a functional block diagram that shows separate digital andanalog signal paths in a receiver. A hybrid radio broadcast signal isreceived on antenna 52, and is converted to a digital signal in ADC 68.The hybrid signal is then split into a digital signal path 90 and ananalog signal path 92. In the digital signal path 90, the digital signalis acquired, demodulated, and decoded into digital audio samples asdescribed in more detail below. The digital signal spends an amount oftime T_(DIGITAL) in the digital signal path 90, which is a variableamount of time that will depend on the acquisition time of the digitalsignal and the demodulation and decoding times of the digital signalpath. The acquisition time can vary depending on the strength of thedigital signal due to radio propagation interference such as fading andmultipath.

In contrast, the analog signal (i.e., the digitized analog audiosamples) spends an amount of time T_(ANALOG) in the analog signal path92. T_(ANALOG) is typically a constant amount of time that isimplementation dependent. It should be noted that the analog signal path92 may be co-located with the digital signal path on the basebandprocessor 82 or separately located on an independent analog processingchip. Since the time spent traveling through the digital signal pathT_(DIGITAL) and the analog signal path T_(ANALOG) may be different, itis desirable to align the samples from the digital signal with thesamples from the analog signal within a predetermined amount so thatthey can be smoothly combined in the audio transition module 94. Thealignment accuracy will preferably be chosen to minimize theintroduction of audio distortions when blending from analog to digitaland visa versa. The digital and analog signals are combined and travelthrough the audio transition module 94. Then the combined digitizedaudio signal is converted into analog for rendering via thedigital-to-analog converter (DAC) 96. As used in this description,references to “analog” or “digital” with regard to a particular datasample streams in this disclosure connote the radio signal from whichthe sample stream was extracted, as both data streams are in a digitalformat for the processing described herein.

One technique for determining time alignment between signals in digitaland analog pathways performs a correlation between the samples of thetwo audio streams and looks for the peak of the correlation. Timesamples of digital and analog audio are compared as one sample stream isshifted in time against the other. The alignment error can be calculatedby successively applying offsets to the sample streams until thecorrelation peaks. The time offset between the two samples at peakcorrelation is the alignment error. Once the alignment error has beendetermined, the timing of the digital and/or analog audio samples can beadjusted to allow smooth blending of the digital and analog audio.

While the description of the previously existing blend techniqueillustrated in FIG. 3 uses a 1024 sample audio frame used in aparticular audio compression codec, it should be recognized that thetechnique could be applied to 2048 sample audio frames used in othercodecs.

FIG. 4 is a functional block diagram of an apparatus for determiningtiming offset between analog and digital audio streams within a desiredaccuracy using downsampled audio streams. The system of FIG. 4 is morefully described in commonly owned U.S. patent application Ser. No.14/862,800, filed Sep. 23, 2015, which is hereby incorporated byreference. In the embodiment of FIG. 4, the digital signal path 90supplies a first stream of samples representative of the content of thereceived digitally modulated signal on line 100. The samples from thefirst sample stream are stored in buffer 102. The first stream ofsamples on line 104 is filtered by an anti-aliasing filter 106 anddownsampled (decimated) as shown in block 108 to produce a firstdecimated sample stream on line 110. The analog signal path 92 suppliesa second stream of samples representative of the content of the receivedanalog modulated signal on line 112. Samples from the second stream ofsamples are stored in buffer 114. The second stream of samples on line116 is filtered by an anti-aliasing filter 118 and downsampled(decimated) as shown in block 120 to produce a second decimated samplestream on line 122. A correlator 124 performs a cross-correlation onsamples of the first and second decimated sample streams and a peakdetector 126 determines an offset between samples of the decimatedstreams that are most highly correlated. Due to the decimation of theinput signals, the peak detector output actually represents a range ofpossible stream offsets. This offset range is then used to determine ashift value for one of the first and second sample streams, asillustrated in block 128. Then the shifted sample stream is decimatedand correlated with the decimated samples from the unshifted stream. Byrunning the estimation multiple times with a shifted input, the range ofvalid results is now limited to the intersection of the range of validresults of the first estimation and the range of valid results of thesecond estimation. The steps of shifting, decimating, correlating andpeak detection can be repeated until a desired accuracy of the timealignment of the first and second sample streams is achieved. At thatpoint, a control signal is output on line 130. Then the blend control132 can use the control signal to blend the analog and digital signalpaths.

The correlation operation performed by the correlator may includemultiplying together decimated data from each stream. The result of themultiplication may appear as noise, with a large peak when the datastreams are aligned in time.

In the system of FIG. 4, the peak detector may analyze correlationresults over time to search for peaks that indicate that the digitaldata streams are aligned in time. In some embodiments, a squaringfunction may square the product output by the correlator in order tofurther emphasize the peaks. Based on the received data, the peak searchunit may output an indication of the relative delay between the analogdata stream and the digital data stream. The indication of relativedelay may include an indication of which one of the two data streams isleading the other.

Once the analog and digital data streams are sufficiently aligned, ablend operation may begin. The blend operation may be conducted, forexample, by reducing the contribution of the analog data stream to theoutput audio while correspondingly increasing the contribution of thedigital data stream until the latter is the exclusive source.

The transition time between the analog and digital audio outputs isgenerally less than one second, which is limited by the diversity delayand receiver decoding times. The relatively short blend transition timepresents challenges in designing blending systems. It has been observedthat frequent transitions between the analog and digital audio can besomewhat annoying when the difference in audio quality and loudnessbetween the digital audio and the analog audio is significant. This isespecially significant when the digital signal has a wider audiobandwidth than the analog audio, and the digital signal is stereo whilethe analog is mono. This phenomenon can occur in mobile receivers infringe coverage areas when highway overpasses (or power lines for AM)are frequently encountered.

International Telecommunication Union Recommendation ITU-R BS.1770-3specification, hereinafter referred to as ITU 1770, is a primarystandard for loudness measurement. ITU 1770 algorithms can be used tomeasure audio program loudness and true-peak audio level. In ITU 1770,the Equivalent Sound Level, L_(eq), is simply defined as the RMS soundpower of the signal relative to a reference sound power. Thiscalculation is easily accomplished with minimal memory and MIPS(millions of instructions per second). An optional frequency weightingprior to the sound power calculation is specified as an “RLB” filter,which is a simple low pass at ˜100 Hz followed by a filter that appliesa 4 dB boost to frequencies above approximately 2 kHz. Adding the filtercalculations for an RLB weighting filter does not require significantlymore MIPS/Memory.

The loudness difference between analog and digital audio can changedynamically. For example, up to 10 dB in loudness difference has beenmeasured when comparing analog and digital audio at various points inthe same program. If the loudness difference is small when blending todigital and later in the program the difference becomes greater,possibly 10 dB greater, a blend back to analog would result in anunacceptable abrupt change in loudness. This is primarily due to thedynamic nature of the loudness difference between digital and analogaudio within a single program. This loudness difference exists fornumerous reasons including, but not limited to, different processingapplied to the analog and digital audio, poor signal conditions, etc.

A short term loudness match at the time a blend operation is performed,coupled with a long term loudness equalization of the digital audio cansolve this fundamental problem.

There are conflicting requirements when setting analog and digitalloudness in the HD Radio system. The first requirement, referred to as a“long term loudness difference”, requires that the loudness perceivedover the duration of the program must be consistent whether listening tothe analog stream or the digital stream. The second requirement,referred to as a “short term loudness difference”, occurs at thetransition time between the two streams. This transition time isgenerally short (e.g., <1 second), and the loudness must be relativelyequal (e.g., ±2 dB), or else the listener will perceive the difference.Measurements have found that the short and long term loudness values canbe drastically different as the content of the program changes.Therefore, at the point of blend a short term value is used so that thetransition time sounds smooth. The short term loudness can be determinedover a short time interval. The short time interval is a time in therange of 1 to 5 seconds. In one embodiment, the short time interval is2.97 seconds. The ideal short time interval for a particular applicationcan be determined based on audio perception and perceptual memory, suchas what is perceived by human hearing to be instantaneous.

The short term loudness value can be slowly ramped to the long termvalue as the program continues so that the overall perceived loudness ofa given program is the same regardless of whether the analog or digitalaudio stream is playing. The long term loudness can be determined over along time interval. The long term loudness can be determined over a longtime interval. The long time interval is a time in the range of 5 to 30seconds. In one embodiment, the long time interval is in a range of 5.94to 29.72 seconds. The long time interval is always longer than the shorttime interval.

Generally, the short time interval must be several seconds, and alwaysless than the long time interval. In some embodiments, the long timeinterval is measured in integer multiples of the short time interval.This is not a strict requirement for the process, but was chosen tosimplify implementation.

When the level of the digital audio stream matches the level of theanalog audio stream, the streams can be blended to produce an audiooutput signal. The short term loudness measurement is calculated andused to update a long term running average loudness value. The minimumtime before blend may occur when level control is enabled is the shorttime interval.

FIG. 5 is a flow block diagram of a method for level alignment inaccordance with an embodiment of the invention. The process starts byreceiving audio samples over a short time interval as shown in block150. The analog and digital signal streams are separated as shown inblocks 152 and 154. This separation may be accomplished using theelements of FIG. 3. The analog and digital audio signals may be filteredusing a perceptual filter to improve the results.

A short term average power (loudness) is calculated for each stream asshown in blocks 156 and 158. This calculation can be performed using thealgorithm set forth in ITU-1770. Then a short term average gain iscalculated as shown in block 160. The short term average gain iscalculated as the linear ratio of analog audio power to digital audiopower. Block 162 shows that the long term average gain is thencalculated, either directly or using the short term gain. The short termaverage gain is the gain determined over the short time interval. Thelong term average gain is the gain determined over the long timeinterval.

The next step depends on whether or not the long time interval has beenmet as shown in block 164. In one implementation, the long time intervalis comprised of integer multiples of the short time interval and thelong term gain is calculated using a running average of the short termgain. Another implementation could calculate the short term and longterm gain independently over different intervals. An audio frame countercan be used to determine when each of the short and long time intervalshas been met.

If the long time interval has been met, the long term gain (runningaverage) over the full long time interval is used as shown in block 166.If the long time interval has not been met, the short term gain is usedas shown in block 168. The short term gain may be averaged withpreviously calculated short term gain measurements to generate a partiallong term gain, but this is not a strict requirement. In either case,the gain is converted from a linear ratio to integer dB (always roundingdown), as shown in block 170, and provided to a host processor for thepurpose of adjusting the digital audio loudness during blend to bettermatch the loudness of the analog audio. The range of digital gaincorrection which can be applied is −8 dB to 7 dB, in 1 dB increments.

The next step depends on whether or not the output of the receiver hasalready been blended to digital as shown in block 172. If the output ofthe receiver has already been blended to digital, the digital gain isadjusted by a predetermined amount (e.g., 1 dB) towards the calculatedlong term gain, as shown in block 174. The adjustment step size shouldbe less than 1.5 dB to avoid immediately perceptible changes in outputvolume. If the output of the receiver has not been blended to digital,the digital gain is set to the calculated gain, as shown in block 176.The updated digital gain parameter is provided to an external audioprocessor, as shown in block 178. Then the short time interval is endedas shown in block 180 and a new short time interval is used forsubsequent iterations of the process as shown in block 150.

The method illustrated in FIG. 5 periodically provides a gain parameterwhich can be used by the host to adjust the digital stream loudness sothat over the long term digital loudness matches long term analogloudness. Before each blend to digital a short termmeasurement/adjustment of a minimum interval can be used to guaranteethe loudness difference at the blend point is minimal. From that pointon a continuous measurement of the analog and digital audio streamlevels could be made using a known type of loudness measurement, such asthe technique defined in the ITU-1770. The loudness measurements arealways performed on the original digital audio stream provided from thedecoder. The digital audio gain is a parameter which is sent to theexternal audio processor on the host. The host is responsible forapplying the loudness adjustment to the output audio. This system doesnot directly modify the analog or digital audio in any way.

Updating the gain of the digital audio signal with this long termloudness difference value could drive the long term average loudness ofthe digital to match that of the analog. If the step size were keptsmall, for example 1 dB, and the update rate were sufficiently long, forexample 3, 5 or 10 seconds, then the difference in audio level could beimperceptible to a listener. After a time the loudness measurementswould stabilize and digital volume would reliably track the analogvolume. This would minimize the potential volume difference at the nextblend to analog without causing major changes in the digital volumeduring playback.

In one embodiment, the short term level measurement can be performed onsamples that occur after time alignment resulting in a longer delay toblend. However, the time alignment algorithm can be run multiple timesto ensure consistency. Then the short term level alignment function canbe run concurrently with a second (or subsequent) execution of the timealignment algorithm, using the alignment value from the first execution.In addition, because the short term level alignment can be executedseparately from time alignment, the level alignment algorithm could berun continuously (for example over a 3 second sample window) regardlessof the time alignment range.

The functions shown in FIG. 5 can be implemented in the circuitry of aradio receiver, using for example, one or more processors that areprogrammed or otherwise configured to perform the functions describedherein. Other hardware embodiments, as well as software embodiments andcombinations thereof may also be used to implement the describedmethod(s).

While the present invention has been described in terms of its preferredembodiments, it will be apparent to those skilled in the art thatvarious modifications can be made to the described embodiments withoutdeparting from the scope of the invention as defined by the followingclaims.

What is claimed is:
 1. A method for processing a digital an audiobroadcast signal including multiple audio streams using an audio signalreceiver, the method comprising: (a) separating an analog audio portiona first audio sample stream of the digital audio broadcast signal from adigital audio portion second audio sample stream of the digital audiobroadcast signal using the audio signal receiver; (b) determining aloudness of the analog audio portion first audio sample stream over afirst short time interval; (c) determining a loudness of the digitalaudio portion second audio sample stream over the first short timeinterval; (d) using the loudness of the analog audio portion first audiosample stream and the loudness of the digital audio portion second audiosample stream to calculate a short term average gain; (e) determining along term average gain; (f) converting one of the long term average gainor the short term average gain to dB decibels (dB) and providing theconverted average gain to an audio processor; (g) if an output of areceiver has been blended to digital, performing one of blending anoutput of a receiver to the second audio sample stream and adjusting adigital second audio sample stream gain parameter toward the long termaverage gain by a preselected increment, to produce an adjusted digitalgain parameter; (h) if the output of the receiver has not been blendedto digital or not blending the output of the receiver to the secondaudio sample stream and setting the digital second audio sample streamgain parameter to the short term average gain; (i)(h) providing thedigitalsecond audio sample stream gain parameter adjusted byfrom step(g) or step (h) to the audio processor; and (j)(i) repeating steps (a)through (i)(h) using a second short time interval.
 2. The method ofclaim 1, wherein: the short term average gain is a linear ratio ofanalog audio power of the first audio sample stream and digital audiopower of the second audio sample stream.
 3. The method of claim 1,wherein: the digital second audio sample stream gain parameter isadjusted toward the short term average gain; and the preselectedincrement is 1 dB.
 4. The method of claim 1, wherein: the long termaverage gain comprises a running average of the short term average gain.5. The method of claim 1, wherein: the long term average gain isdetermined independently of the short term average gain.
 6. The methodof claim 1, wherein: the long term average gain is converted to dB if along time interval has been met and the short term average gain isconverted to dB if the long time interval has not been met.
 7. Themethod of claim 6, wherein the long time interval comprises an integermultiple of the short time intervals.
 8. The method of claim 6, wherein:the short time interval is in a range from 1 to 5 seconds and the longtime interval is in a range from 5 to 30 seconds.
 9. The method of claim1, wherein: the analog audio portion first audio sample stream comprisesa stream of samples analog modulated program material; and the digitalaudio portion second audio sample stream comprises a stream of samplesdigitally modulated program material.
 10. The method of claim 1, whereinthe measurements of the levels of the analog audio portion first audiosample stream and the digital audio portion second audio sample streamare performed in accordance with the International TelecommunicationsUnion Recommendation (ITU-R) BS.1770 specification.
 11. A radio An audiosignal receiver comprising: processing circuitry configured to: (a)separate an analog a first audio portion sample stream of the digital anaudio broadcast signal from a digital second audio portion sample streamof the digital audio broadcast signal; (b) determine a loudness of theanalog first audio portion sample stream over a first short timeinterval; (c) determine a loudness of the digital second audio portionsample stream over the first short time interval; (d) use the loudnessof the analog first audio portion sample stream and the loudness of thedigital second audio portion sample stream to calculate a short termaverage gain; (e) determine a long term average gain; (f) convert one ofthe long term average gain or the short term average gain to dB decibels(dB) and provide the converted average gain to an audio processor; (g)if an output of a receiver has been blended to digital, perform one ofblend an output of a receiver to the second audio sample stream andadjust a digital second audio sample stream gain parameter by apreselected increment to produce an adjusted digital gain parameter; (h)if the output of the receiver has not been blended to digital, or notblend the output of the receiver to the second audio sample stream andset the digital second audio sample stream gain parameter to the shortterm average gain; (i)(h) provide the digitalsecond audio sample streamgain parameter adjusted byfrom step (g) or step (h) to the audioprocessor; and (j)(i) repeat steps (a) through (i)(h) using a secondshort time interval.
 12. The radio audio signal receiver of claim 11,wherein: the short term average gain is a linear ratio of analog audiopower and digital audio power.
 13. The radio audio signal receiver ofclaim 11, wherein: the digital second audio sample stream gain parameteris adjusted toward the short term average gain; and the preselectedincrement is 1 dB.
 14. The radio audio signal receiver of claim 11,wherein: the long term average gain comprises a running average of theshort term average gain.
 15. The radio audio signal receiver of claim11, wherein: the long term average gain is determined independently ofthe short term average gain.
 16. The radio audio signal receiver ofclaim 11, wherein: the long term average gain is converted to dB if along time interval has been met and the short term average gain isconverted to dB if the long time interval has not been met.
 17. Theradio audio signal receiver of claim 16, wherein the long time intervalcomprises an integer multiple of the short time intervals.
 18. The radioaudio signal receiver of claim 16, wherein: the short time interval isin a range from 1 to 5 seconds and the long time interval is in a rangefrom 5 to 30 seconds.
 19. The radio audio signal receiver of claim 11,wherein: the analog first audio portion sample stream comprises a streamof samples analog modulated program material; and the digital secondaudio portion sample stream comprises a stream of samples digitallymodulated program material.
 20. The radio audio signal receiver of claim11, wherein the measurements of the levels of the analog first audioportion sample stream and the digital second audio portion sample streamare performed in accordance with the International TelecommunicationsUnion Recommendation (ITU-R) BS.1770 specification.